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Summary:ASTERISK-23010: No BYE message sent when sip INVITE is received
Reporter:Ryan Tilton (intelafone)Labels:
Date Opened:2013-12-16 15:05:56.000-0600Date Closed:2014-01-14 12:45:03.000-0600
Priority:CriticalRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:11.5.0 Frequency of
Occurrence
Related
Issues:
Environment:CentosAttachments:
Description:I am not sure if this is a bug but it was a crucial step in getting Shared call appearances working for me.  

In chan_sip.c under handle_invite_replaces function if you remove

ast_setstate(c, AST_STATE_DOWN);

right before the

ast_hangup(c);

This causes the bye message to be sent out correctly to the UA that is being replaced.  



Comments:By: Rusty Newton (rnewton) 2013-12-17 11:44:57.262-0600

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. the specific steps or actions you took that caused you to encounter the problem,
2. the behavior you expected, and
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!



By: Rusty Newton (rnewton) 2013-12-17 11:46:26.509-0600

Please attach a packet capture of a reproduction, and after your fix. Include an Asterisk log following the instructions here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

By: Rusty Newton (rnewton) 2014-01-07 09:58:14.224-0600

Mark Michelson looked at this confirmed it is a legitimate bug. Putting this into open state.